In the field of telephony communication, there have been many technological advances in technology over the years that have contributed to more efficient use of telephone communication within hosted call-center environments. Most of these improvements involve integrating the telephones and switching systems in such call centers with computer hardware and software adapted for, among other things, better routing of telephone calls, faster delivery of telephone calls and associated information, and improved service with regards to client satisfaction. Such computer-enhanced telephony is known in the art as computer-telephony integration (CTI).
Generally speaking, CTI implementations of various design and purpose are implemented both within individual call-centers and, in some cases, at the telephone network level. For example, processors running CTI software applications may be linked to telephone switches, service control points (SCP), and network entry points within a public or private telephone network. At the call-center level, CTI-enhanced processors, data servers, transaction servers, and the like, are linked to telephone switches and, in some cases, to similar CTI hardware at the network level, often by a dedicated digital link. CTI and other hardware within a call-center is commonly referred to as customer premises equipment (CPE). It is the CTI processor and application software in such centers that provides computer enhancement to a call center.
In a CTI-enhanced call center, telephones at agent stations are connected to a central telephony switching apparatus, such as an automatic call distributor (ACD) switch or a private branch exchange (PBX). The agent stations may also be equipped with computer terminals such as personal computer/video display units (PC/VDUs) so that agents manning such stations may have access to stored data as well as being linked to incoming callers by telephone equipment. Such stations may be interconnected through the PC/VDUs by a local area network (LAN). One or more data or transaction servers may also be connected to the LAN that interconnects agent stations. The LAN is, in turn, connected to the CTI processor, which is connected to the call switching apparatus of the call center.
In recent years, further advances in computer technology, telephony equipment, and infrastructure have provided many opportunities for improving telephone service in publicly switched and private telephone intelligent networks. Similarly, development of a separate information and data network known as the Internet, together with advances in computer hardware and software have led to a new multimedia telephone system known in the art by several names. In this new systemology, telephone calls are simulated by multi-media computer equipment, and data, such as audio data, is transmitted over data networks as data packets. In this application the broad term used to describe such computer-simulated telephony is Data-Network Telephony (DNT).
A typical DNT system is not a dedicated or connection oriented system. That is, data, including audio data, is prepared, sent, and received as data packets. The data packets share network links, and may travel by varied and variable paths. There is thus no generally dedicated bandwidth, unless special systems, such as RSVP systems known in the art, are used for guaranteeing bandwidth during a call. DNT calls must share the bandwidth available on the network in which they are traveling.
Recent improvements to available technologies associated with the transmission and reception of data packets during real-time DNT communication have made it possible to successfully add DNT, principally, Internet Protocol Network Telephony (IPNT) capabilities to existing CTI call centers. Such improvements, as described herein and known to the inventors, include methods for guaranteeing and verifying available bandwidth or quality of service (QoS) for a transaction, improved mechanisms for organizing, coding, compressing, and carrying data more efficiently using less bandwidth, and methods and apparatus for intelligently replacing lost data via using voice supplementation methods and enhanced buffering capabilities.
In typical call centers, DNT is accomplished by Internet connection and IPNT calls. In systems known to the inventors, incoming IPNT calls are processed and routed within an IPNT-capable call-center in much the same way as COST calls are routed in a CTI-enhanced center, using similar or identical call models. A call model is essentially a set of services and logic provided for enabling call routing, switching, and so on.
Call centers having both CTI and IPNT capability utilize LAN-connected agent-stations with each station having a telephony-switch-connected headset or phone, and a PC connected, in most cases via LAN, to the network carrying the IPNT calls. Therefore, in most cases, IPNT calls are routed to the agent's PC while conventional telephony calls are routed to the agent's conventional telephone or headset.
Companies have, for some time, experimented with various forms of integration between the older COST systems and newer IPNT systems. For example, by enhancing data servers, interactive voice response units (IVR), agent-connecting networks, and so on, with the capability of understanding Internet protocol, data arriving from either network may be integrated requiring less equipment and lines to facilitate processing, storage, and transfer of data.
In a network system known to the inventors and described with reference to Ser. No. 09/024,923, listed in the Cross-Reference section, a computerized telephony bridge unit maintained in the network has a Data Network Telephony (DNT) Port and a Connection Oriented/Switched Telephony (COST) trunk port. Each port is associated with circuitry for receiving and placing calls in the data format required by the connected networks. The bridge unit further comprises conversion circuitry for converting data dynamically between network protocols compatible with each connected network.
In this system, control routines are provided and are executable on the computerized bridge unit. The control routines are adapted to receive a first call from one of the COST or DNT networks, to place a call associated with the received call on the network other than the network on which the call is received, and to dynamically convert data between a call connected at one port and a call connected at the other port. The data network can be the Internet, and the COST network can be any publicly or privately switched dedicated-connection-oriented telephone network.
One with skill in the art will recognize that there are several Internet protocols, CTI protocols, and Device protocols, which have been proposed and adopted as standard or semi-standard protocols for streamlining integrated telephony between disparate networks. For example, an Internet protocol known as H.323 is a standard approved by the International Telecommunication Union (ITU) that defines how audiovisual conferencing data is transmitted across networks. In theory, H.323 should enable users to participate in a same telephony conference even though they are using different videoconferencing applications. Although most videoconferencing vendors maintain that their products conform to H.323, such adherence may not actually produce seamless inter-operability.
Another known protocol termed Media Gateway Control Protocol (MGCP) was developed by Telcordia™ in cooperation with Level 3 Communications™. This protocol is an internal protocol which is was developed to work with existing signaling protocols such as H.323, SIP or SS7. One reason new standards are being developed is because of the growing popularity of what is termed Voice over IP (VoIP ).
Standard telephones are relatively inexpensive because they are not complex in terms of intelligence. Standard telephones work with specific switches at some central switching apparatus, if they happen to be so connected. IP telephones and devices, on the other hand, are not fixed to a specific switch, so they must contain processors that enable them with the operating intelligence that is independent from a central switching location (no third party control). MGCP is meant to simplify standards for this new technology by eliminating the need for complex, processor-intense IP telephony devices by providing some third party control, thus simplifying and lowering the cost of these end terminals.
A protocol representing a basic telephony call model is known to the inventors as Computer-Supported Telephony Applications (CSTA). ECMA is the international standards organization that defined the CSTA resource model and protocol. To connect a telephone system to Computer Telephony (CT) Connect, a telephone system vendor must provide a CSTA-compliant, ASN.1 encoded message flow. This can be provided across a number of different transports, but TCP/IP is becoming the most popular.
Although the developed protocols do much to facilitate seamless communication between networks adding some third party call control, it becomes apparent that third party control over telephony practiced in VoIP applications is severely limited. Proposed prior-art solutions using developed and hopefully standardized protocols in addition to the provision of special gateways has added complexity more than simplicity for enterprises attempting to integrate CTI telephony regimens into VoIP and other data packet venues.
Arguably, the most frustrating of these challenges is providing consistent call model representation both at the call control entity (CCE) and the switching entity (SWE). Traditionally, CTI protocols for dedicated networks have been largely vendor-specific, forcing CTI software vendors to develop separate software modules for each switch model. The above-mentioned protocols were proposed and developed as potential standards for CTI links.
Unfortunately, attempts of such CTI protocol standardization are not likely to produce compatible implementations. It is easy to see that any non-trivial CTI software suite has a need to maintain an accurate replica of the switch state, which in practice means that the CTI software has to replicate the call model of a particular switch.
Any discrepancy between an actual call model implemented by a switch vendor and its reverse-engineered replica in CTI control software causes loss of coherency between the actual switching state and its image in the control software. Moreover, practically all switch vendors introduce subtle changes to their call model in successive versions of switch software (this is unavoidable when new features are added and programming errors are corrected). Packet telephony makes utilization of call models even more complicated by replacing centralized switches with a heterogeneous, distributed switching environment, multiplying the effects of programming errors, revision levels etc.
What is clearly needed is a low-level protocol that enables negotiation over a network between a CCE having service logic and a SWE providing only switching functions such that only one call model is required and SWE functions may be implemented according to attributes of the model.
Furthermore, it is desirable, for such a protocol to run on as many devices, systems etc. without much adaptation.